Live Migration of Virtualized Carrier Grade SIP Server

The concept of network virtualization, such as network functions virtualization, has attracted considerable attention from telecom carriers and a live migration technique is a key feature of virtualization technology. However, there are some challenges associated with applying server virtualization technology including live migration to a SIP server. Previous work has not dealt with the performance or behavior of a SIP server during live migration. Neither has it targeted a carrier grade SIP server for live migration. In this paper we present a virtualized carrier grade SIP server running on a virtual machine, which is configured with Carrier Grade Linux, HA middleware and SIP-AS application. We also assess its performance to investigate the impact of throughput degradation and suspension on a SIP layer and HA cluster configuration

Agent based infrastructure for real-time applications

In this paper we propose a new infrastructure for real-time applications. As a preliminary, we describe basic characteristics of the most popular real-time services like VoIP, videoconferencing, live media streaming, and network multiplayer games. We focus on the end-to-end latency, bandwidth and efficient transmission methods. Next, we present our project concepts, infrastructure model, details of implementation and our testing environment which was designed for testing many aspects of real-time services. The system combines mechanisms for ensuring best possible connection quality (QoS), load balance of servers in infrastructure and gives control over the packet routing decisions. Additionally, provided security mechanisms make it a good choice even in the environment where a high security level is required. The system is based on the Peer-to-Peer (P2P) model and data between users is routed over an overlay network, consisting of all participating peers as network nodes. This overlay can by used for application level multicast or live media stream. In the logging process each user is assigned to a specific node (based on his geographic location and nodes load). Because nodes are participating in data transmission, we have control over the data flow route. It is possible to specify the desired route, so, regardless of the external routing protocol, we can avoid paths that are susceptible to eavesdropping. Another feature of the presented system is usage of agents. Each agent acts within the single node. Its main task is to constantly control the quality of transmission. It analyzes such parameters like link bandwidth use, number of lost packets, time interval between each packet etc. The information collected by the agents from all nodes allows to build a dynamic routing table. Every node uses the Dijkstra's algorithm to find the best at the moment route to all other nodes. The routes are constantly modified as a consequence of changes found by agents or updates sent by other nodes. In VoD services agents also analyze popularity of streamed media, which helps build intelligent video cache. To ensure greater security and high reliability of the system, we have provided a reputation mechanism. It is used during bringing up to date the information about possible routes and their quality, given by other nodes. Owing to this solution nodes and routes which are more reliable get higher priority

VoIP Performance over Different Interior Gateway Protocols

Voice over IP technology integrates data and voicenetworks and offers flexibility by supplying device interoperabilityusing standards-based protocols. Routing is an essential datanetworking function that provides an efficient real-time datadelivery VoIP requires. Best-effort networks leverage InteriorGateway Protocol technologies to determine paths for routingpackets between hosts. Architects for best-effort networks provideservices by over-provisioning the links and routers so that networkcongestion does not introduce unwanted levels of latency, jitter, andpacket loss. Route re-convergence can be detrimental to VoIP usersin the middle of a conversation, as it impacts both latency and jitter.This paper studies how VoIP performance can be affected bydifferent routing behaviors which include Routing InformationProtocol, Open Shortest Path First and Enhanced Interior GatewayRouting Protocol. Network modeling and simulation have beencarried out with OPNET Modeler to evaluate and compareperformances